CN1776805B - Low internal-memory-demand digital reverberation system and method - Google Patents
Low internal-memory-demand digital reverberation system and method Download PDFInfo
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- CN1776805B CN1776805B CN 200410091092 CN200410091092A CN1776805B CN 1776805 B CN1776805 B CN 1776805B CN 200410091092 CN200410091092 CN 200410091092 CN 200410091092 A CN200410091092 A CN 200410091092A CN 1776805 B CN1776805 B CN 1776805B
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Abstract
The system includes an adder, unit for descending frequency, a compression set, a delay time register, an expanding device, an attenuating device, and interpolation unit. The unit for descending frequency carries out calculation for lowered sampling rate for an audio data stream. The compression set compresses data stream in lowered frequency. Storing the compressed data stream momentarily, the delay time register outputs the compressed data stream after a delayed time according to a prearranged delay time. The expanding device carried out operation of decompression for the delayed compressed data. The attenuating device attenuates the decompressed data. The interpolation unit carries out operation of interpolation for the attenuated data stream in order to increase data quantity. The adder adds the interpolated audio data stream to audio data stream to make the audio data possess echo effect.
Description
Technical field
The present invention relates to a kind of digital reverberation System and method for, refer to a kind of digital reverberation System and method for of low memory requirements especially.
Background technology
Because the fast development of multimedia technology, all kinds of multimedia devices (for example: DVD videocorder, digital camera, digital VTR, Karaoke home audio-visual system) have goed deep into each family.Reverberation is the function of Karaoke home audio-visual VS .NET Prerequisites, can allow the sound signal of being imported by microphone increase reverberation (echo) effect.Fig. 1 is the calcspar of known generation reverberation (echo) system, and it comprises a low-pass filter 110, a deferred mount 120, an attenuating device 130 and an adder 140.At first a sound signal is passed through this low-pass filter 110,, reduce the frequency range of this sound signal through after these low-pass filter 110 low-pass filtering.Sound signal after these deferred mount 120 storage low-pass filtering.The sound signal that this attenuating device 130 will postpone output is decayed, and adder 140 is with the sound signal of decay and original sound signal addition, to produce a sound signal with echo (echo) effect.
Yet when echo (echo) effect need have 0.1 second delay, one 8 sound signals are taken a sample with the sampling rate of 44.1K, the sample number of 4.41K is arranged, that is this deferred mount 120 need have the 4.41K byte, can reach 0.1 second delay.The stored memory that this cost is many.At above-mentioned shortcoming, United States Patent (USP) USP5,917, No. 917 bulletin exposure are in the mode of sample rate conversion (sampling rate conversion), earlier speech data is carried out reduced sampling (Downsampling) computing, to reduce data volume, to reduce the use amount of internal memory.United States Patent (USP) USP6,647, No. 064 bulletin exposure is in the mode of adaptive differential pulse code modulation system (AdaptiveDifferential Pulse Code Modulation, ADPCM), after voice signal compressed, data storage after will compressing again is in internal memory, to reduce the use amount of internal memory.Though the ADPCM compressibility is than higher, ADPCM compression and the coding and decoding that decompresses are complicated, and because ADPCM can't also increase the complexity of control from beginning decoding more arbitrarily.So known digital reverberation System and method for still has the space of improvement.
Summary of the invention
An object of the present invention is to provide a kind of digital reverberation System and method for of low memory requirements, can reduce the data volume of audio data stream, to reduce the use amount of internal memory.
Another object of the present invention provides a kind of digital reverberation System and method for of low memory requirements, can't be can avoid known technology to use the ADPCM computing to be produced from the problem that a bit begins arbitrarily to decipher, and the complexity of control when also reducing coding and decoding.
According to one aspect of the present invention, proposing a kind of digital reverberation system of low memory requirements, mainly comprise a totalizer, a lowering freqyency device, a compression set, a delay time register, an extension apparatus, an attenuating device and an interpolating device. the first input end of this totalizer is in order to import an audio data stream; This lowering freqyency device is connected to the output terminal of this totalizer, reduces the sampling rate computing with the audio data stream to this totalizer output, obtains a frequency reducing data stream; This compression set is connected to this lowering freqyency device, so that this frequency reducing data stream is compressed processing, produces packed data stream; This delay time register is connected to this compression set, with temporary this packed data stream, and according to a predetermined time delay, postpones this packed data stream of output; This extension apparatus is connected to this delay time register, so that the packed data stream that postpones is carried out decompression operation; This attenuating device is connected to this extension apparatus, so that the decompressed data stream of its output is decayed; This interpolating device is connected to this attenuating device, carries out interpolation operation with the attenuation data stream to this attenuating device output, to increase data volume; Wherein, this totalizer second input end is connected to this interpolating device, so that the sound signal that this interpolating device was produced is added in this audio data stream, and makes this audio data stream have echo (echo) effect.
According to another aspect of the present invention, propose a kind of digital reverberation system of low memory requirements, mainly comprise a first adder, a lowering freqyency device, a compression set, a delay time register, a plurality of extension apparatus, a plurality of attenuating device, a second adder and an interpolating device.The first input end of this first adder is in order to import an audio data stream; This lowering freqyency device is connected to the output terminal of this first adder, reduces the sampling rate computing with the audio data stream to this first adder output, obtains a frequency reducing data stream; This compression set is connected to this lowering freqyency device, so that this frequency reducing data stream is compressed processing, to produce packed data stream; This delay time register is connected to this compression set, with temporary this packed data stream, and has multistage delay output terminal, to postpone this packed data stream of output; These a plurality of extension apparatus are connected to the multilevel delay output terminal of this delay time register respectively, so that the packed data stream that postpones is carried out decompression operation; These a plurality of attenuating devices are connected to this a plurality of extension apparatus, so that the decompressed data stream of its output is decayed; This second adder has a plurality of input ends, and these a plurality of input ends are connected to this a plurality of attenuating devices, carries out additive operation with a plurality of attenuation data streams to these a plurality of attenuating device outputs; This interpolating device is connected to this second adder output terminal, so that the attenuation data stream after adding up is carried out interpolation operation, to increase data volume; Wherein, this first adder second input end is connected to this interpolating device, so that the sound signal that this interpolating device was produced is added in this audio data stream, and makes this audio data stream have echo (echo) effect.
Propose a kind of digital reverberation method of low memory requirements more on the one hand according to of the present invention, this method mainly comprises a frequency reducing step, a compression step, a temporary step, an extension step, an attenuation step, an interpolation step and an addition step.This frequency reducing step receives an audio data stream, and this audio data stream is reduced the sampling rate computing, obtains a frequency reducing data stream; This compression step compresses processing to this frequency reducing data stream; Should flow by temporary this packed data of temporary step system, and, export this packed data stream according to a predetermined time delay; This extension step system carries out decompression operation to the packed data stream of this delay output, to produce decompressed data stream; This attenuation step is decayed to this decompressed data stream, to produce attenuation data stream; This interpolation step is carried out interpolation operation to this attenuation data stream, increases data volume, to produce interpolation data stream; This addition step flows this interpolation data in this audio data stream that adds in this frequency reducing step, and makes this audio data stream have echo (echo) effect.
Description of drawings
Fig. 1 is the calcspar of known echogenicity (echo) system.
Fig. 2 is the calcspar of the digital reverberation system of low memory requirements of the present invention.
Fig. 3 is the operation synoptic diagram of lowering freqyency device of the present invention.
Fig. 4 is that the output input of non-linear μ of the present invention-law compaction algorithms concerns synoptic diagram.
Fig. 5 is the fragment linear-apporximation synoptic diagram of the output input relation of non-linear μ of the present invention-law compaction algorithms.
Fig. 6 is the calcspar of delay time register of the present invention.
Fig. 7 is the operation synoptic diagram of interpolating device of the present invention.
Fig. 8 is the calcspar of another embodiment of digital reverberation system of low memory requirements of the present invention.
Fig. 9 is the process flow diagram of the digital reverberation method of low memory requirements of the present invention.
Embodiment
Fig. 2 is the calcspar of the digital reverberation system of low memory requirements of the present invention, mainly comprises a totalizer 210, a lowering freqyency device 220 (decimator), a compression set 230 (compressor), a delay time register 240 (delay buffer), an extension apparatus 250 (expander), an attenuating device 260 (attenuator), reaches an interpolating device 270 (interpolator).The first input end 211 of this totalizer 210 is in order to import an audio data stream.This lowering freqyency device 220 is connected to the output terminal of this totalizer 210, reduces sampling rate computing (decimation) with the audio data stream to these totalizer 210 outputs, obtains a frequency reducing data stream.This compression set 230 is connected to this conversion equipment 220, so that this frequency reducing data stream is compressed processing.
This delay time register 240 is connected to this compression set 230, with the data stream of temporary this compression, and according to a predetermined time delay, postpones the data stream of this compression of output.This extension apparatus 250 is connected to this delay time register 240, so that the packed data stream that postpones is carried out decompression operation.This attenuating device 260 is connected to this extension apparatus 250, so that the decompressed data stream of its output is decayed.This interpolating device 270 is connected to this attenuating device 260, carries out interpolation operation with the attenuation data stream to these attenuating device 260 outputs, to increase data volume.Wherein, these totalizer 210 second input ends 212 are connected to this interpolating device 270, add in this audio data stream with the sound signal that this interpolating device 270 is produced, and make this audio data stream have echo (echo) effect.
Fig. 3 is the operation synoptic diagram of this lowering freqyency device 220, its with audio data stream through low-pass filtering after, carry out a reduced sampling (Downsampling) computing again, to lower the data volume of this audio data stream.Wherein, reducing sampling rate operational factor (decimation factor) M is 3.That is, every interval 3 (=M) individual input sample (sample) sampling, other 2 (=M-1) individual sample all is " discarded ".Therefore, this lowering freqyency device 220 can with this method of taking a sample, be 1 input value with continuous 3 input value abbreviations under the situation that does not influence signal reduction accuracy just, effectively reduces the data volume of this audio data stream.M is 3 to be for purposes of illustration, can not be as limiting interest field of the present invention.
This lowering freqyency device 220 can be realized with a leggy finite impulse response filter (polyphase FIRdecimation filter).Because it can reduce data volume to reduce sampling rate operational factor (decimation factor) M effectively.
230 pairs of these frequency reducing data stream of this compression set are carried out non-linear μ-law compaction algorithms.Fig. 4 is that the output input of a non-linear μ-law compaction algorithms concerns synoptic diagram, and it can use formula (1) expression,
Wherein, μ is one greater than 0 constant, and x is an input signal, and y is an output signal.Fig. 5 is the fragment linear-apporximation synoptic diagram of the output input relation of non-linear μ-law compaction algorithms among Fig. 4,230 output input relations of this compression set according to Fig. 5, this frequency reducing data stream is carried out non-linear μ-law compaction algorithms, required bits number when reducing signal encoding is can further reduce data volume again.
This delay time register 240 can use the flip-flop of N position M level to form, as shown in Figure 6.In Fig. 6, if each pulse bandwidth of clock signal CLK is K microsecond (μ s), 240 of this delay time registers can provide the delay output of M * K microsecond (μ s).This delay time register 240 also can use synchronous first-in first-out register (synchronous first in first out, SFIFO) be achieved, or a first-in first-out register (FIFO) or random access memory (RAM) cooperate read/write circuit to be achieved.
250 pairs of these packed data streams of this extension apparatus carry out an anti-μ-law decompression operation, to revert back the linear PCM value. this attenuating device 260 is decayed the output signal of this extension apparatus 250, mainly be because of signal as echo (echo) effect, its amplitude (amplitude) should be done and be decayed earlier by this attenuating device 260 earlier less than the amplitude of this audio data stream of being imported by this first input end 211.
Generally speaking, the frequency input signal of this lowering freqyency device 220 equates with the signal output frequency of this interpolating device 270.Therefore, through after these lowering freqyency device 220 frequency reducings, must in this interpolating device 270, a plurality of signal interpolations be gone back, with the output signal of acquisition with this interpolating device 270 of the input signal same frequency of this lowering freqyency device 220 at signal.Fig. 7 is the operation synoptic diagram of this interpolating device 270, and it carries out an interpolation (interpolation) computing or oversampling (upsampling) and low-pass filtering computing, to increase the data volume of this audio data stream.Wherein, the interpolation factor (interpolation factor) L is 3.That is, (=L-1) the individual sample that after each input sample (sample), inserts 2.Promptly after each sample, insert 2 (=L-1) individual value is 0 signaling point.This interpolating device 270 can be a leggy finite impulse response filter (polyphase FIRinterpolation filter).
The interpolation audio data stream of the output of this interpolating device 270 via second input end 212 of this totalizer 210, with these totalizer 210 first input ends 211 these audio data stream additions of input, and reaches echo (echo) effect.
Fig. 8 is the calcspar of another embodiment of digital reverberation system of low memory requirements of the present invention, itself and the main difference of Fig. 2 be to increase the N group extension apparatus (851～85N), attenuating device (861～86N).So can produce difference shake good fortune and different deferred echo (echo) effect.
Fig. 9 is the process flow diagram of the digital reverberation method of low memory requirements of the present invention.At first, in step S910, import an audio data stream, this audio data stream can be a pulse code modulated (Pulse Code Modulation, PCM) form.In step S920, carry out a frequency reducing computing, to lower the data volume of this audio data stream.In step S930, carry out a non-linear μ-law compaction algorithms, required bits number when reducing signal encoding is can further reduce data volume again.In step S940, the packed data stream that is produced among the temporary step S930.
In step S950,, packed data stream temporary among the step S940 is read according to a predetermined time delay.In step S960, carry out an anti-μ-law decompression operation, to revert back the linear PCM value.In step S970, carry out a decay computing, to allow as the amplitude (amplitude) of the signal of echo (echo) effect amplitude less than input audio data stream among the step S910.In step S980, carry out an interpolation (interpolation) computing or oversampling (upsampling) computing, increasing the data volume of this audio data stream, to allow the frequency of signal of echo (echo) effect conform to the frequency of input audio data stream among the step S910.In step S990,, has the audio data stream of echo (echo) effect with generation with input audio data stream addition among signal after the interpolation and the step S910.
Length between the adjustment internal memory writes and exports can change time delay, and pad value also can be adjusted and reach different-effect.
In sum, the present invention not only uses reduced sampling (Downsampling) computing, to reduce the data volume of audio data stream, to reduce the use amount of internal memory.Use simultaneously non-linear μ-law compaction algorithms again, required bits number when reducing signal encoding further reduces the data volume of audio data stream.Again, the present invention uses non-linear μ-law compaction algorithms, and can avoid known technology to use the ADPCM computing to produce can't be from the problem that a bit begins arbitrarily to decipher, the complexity of control when also reducing coding and decoding.
The foregoing description only is to give an example for convenience of description, and the interest field that the present invention advocated should be as the criterion so that the claim scope is described certainly, but not only limits to the foregoing description.
Claims (19)
1. the digital reverberation system of a low memory requirements mainly comprises:
One totalizer, its first input end is used for importing an audio data stream;
One lowering freqyency device is connected to the output terminal of this totalizer, reduces the sampling rate computing with the audio data stream to this totalizer output, obtain a frequency reducing data stream, wherein, this lowering freqyency device is carried out a low-pass filtering and a reduced sampling computing, to lower the data volume of this audio data stream;
One compression set is connected to this lowering freqyency device, so that this frequency reducing data stream is compressed processing, to produce packed data stream;
One delay time register is connected to this compression set, with temporary this packed data stream, and according to a predetermined time delay, postpones this packed data stream of output;
One extension apparatus is connected to this delay time register, so that the packed data stream that postpones is carried out decompression operation;
One attenuating device is connected to this extension apparatus, so that the decompressed data stream of its output is decayed; And
One interpolating device is connected to this attenuating device, carries out interpolation operation with the attenuation data stream to this attenuating device output, to increase data volume;
Wherein, this totalizer second input end is connected to this interpolating device, so that the sound signal that this interpolating device was produced is added in this audio data stream, and makes this audio data stream have Echo.
2. the system as claimed in claim 1, wherein, this compression set carries out the non-linear compression computing to this frequency reducing data stream.
3. system as claimed in claim 2, wherein, this non-linear compression computing is one μ-law compaction algorithms.
4. the system as claimed in claim 1, wherein, this extension apparatus flows to the line nonlinearity decompression operation to this packed data.
5. system as claimed in claim 4, wherein, this nonlinear solution compaction algorithms is an anti-μ-law decompression operation.
6. the system as claimed in claim 1, wherein, this interpolating device is carried out an oversampling computing, to increase the data volume of this audio data stream.
7. the digital reverberation system of a low memory requirements mainly comprises:
One first adder, its first input end is in order to import an audio data stream;
One lowering freqyency device is connected to the output terminal of this first adder, reduces the sampling rate computing with the audio data stream to this first adder output, obtains a frequency reducing data stream;
One compression set is connected to this lowering freqyency device, so that this frequency reducing data stream is compressed processing, to produce packed data stream;
One delay time register is connected to this compression set, with temporary this packed data stream, and has multistage delay output terminal, to postpone this packed data stream of output;
More than one extension apparatus is connected to the multilevel delay output terminal of this delay time register respectively, so that the packed data stream that postpones is carried out decompression operation;
More than one attenuating device is connected to this a plurality of extension apparatus, so that the decompressed data stream of its output is decayed;
One second adder, it has a plurality of input ends, and these a plurality of input ends are connected to this a plurality of attenuating devices, carry out additive operation with a plurality of attenuation data streams to these a plurality of attenuating device outputs; And
One interpolating device is connected to this second adder, so that the attenuation data stream after adding up is carried out interpolation operation, to increase data volume;
Wherein, this first adder second input end is connected to this interpolating device, so that the sound signal that this interpolating device was produced is added in this audio data stream, and makes this audio data stream have Echo.
8. system as claimed in claim 7, wherein, this lowering freqyency device is carried out a low-pass filtering and a reduced sampling computing, to lower the data volume of this audio data stream.
9. system as claimed in claim 7, wherein, this compression set flows to the line nonlinearity compaction algorithms to this sampled data.
10. system as claimed in claim 9, wherein, this non-linear compression computing is one μ-law compaction algorithms.
11. system as claimed in claim 7, wherein, this extension apparatus flows to the line nonlinearity decompression operation to this packed data.
12. system as claimed in claim 11, wherein, this nonlinear solution compaction algorithms is an anti-μ-law decompression operation.
13. system as claimed in claim 7, wherein, this interpolating device is carried out an oversampling computing, to increase the data volume of this audio data stream.
14. the digital reverberation method of a low memory requirements, this method mainly comprises the following step:
One frequency reducing step, it receives an audio data stream, and this audio data stream is reduced the sampling rate computing, obtains a frequency reducing data stream, and wherein, this frequency reducing step is carried out a low-pass filtering and a reduced sampling computing, to lower the data volume of this audio data stream;
One compression step, it compresses processing to this frequency reducing data stream;
One temporary step, its temporary this packed data stream, and, export this packed data stream according to a predetermined time delay;
One extends step, and its packed data stream to this delay output carries out decompression operation, to produce decompressed data stream;
One attenuation step, it is decayed to this decompressed data stream, to produce attenuation data stream;
One interpolation step, it carries out interpolation operation to this attenuation data stream, increases data volume, to produce interpolation data stream; And
One addition step in this audio data stream in this this frequency reducing step of interpolation data stream adding, and makes this audio data stream have Echo.
15. method as claimed in claim 14, wherein, this compression step carries out the non-linear compression computing to this frequency reducing data stream.
16. method as claimed in claim 15, wherein, this non-linear compression computing is one μ-law compaction algorithms.
17. method as claimed in claim 14, wherein, this extension step flows to the line nonlinearity decompression operation to this packed data.
18. method as claimed in claim 17, wherein, this nonlinear solution compaction algorithms is an anti-μ-law decompression operation.
19. method as claimed in claim 14, wherein, this interpolation step is carried out an oversampling computing, to increase the data volume of this audio data stream.
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Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1074795A (en) * | 1992-01-24 | 1993-07-28 | 郎武株式会社 | Echo-adding circuit |
| CN1191659A (en) * | 1995-06-01 | 1998-08-26 | 艾利森电话股份有限公司 | Arrangement and method in mobile communication system for selecting communication arrangement |
| EP1432222A1 (en) * | 2002-12-20 | 2004-06-23 | Siemens Aktiengesellschaft | Echo canceller for compressed speech |
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| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1074795A (en) * | 1992-01-24 | 1993-07-28 | 郎武株式会社 | Echo-adding circuit |
| CN1191659A (en) * | 1995-06-01 | 1998-08-26 | 艾利森电话股份有限公司 | Arrangement and method in mobile communication system for selecting communication arrangement |
| EP1432222A1 (en) * | 2002-12-20 | 2004-06-23 | Siemens Aktiengesellschaft | Echo canceller for compressed speech |
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